| /* |
| * Copyright 2013 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <stdint.h> |
| |
| #include <cstddef> |
| #include <limits> |
| #include <memory> |
| #include <optional> |
| #include <string> |
| #include <type_traits> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/strings/match.h" |
| #include "api/audio_codecs/L16/audio_decoder_L16.h" |
| #include "api/audio_codecs/L16/audio_encoder_L16.h" |
| #include "api/audio_codecs/audio_codec_pair_id.h" |
| #include "api/audio_codecs/audio_decoder.h" |
| #include "api/audio_codecs/audio_decoder_factory.h" |
| #include "api/audio_codecs/audio_decoder_factory_template.h" |
| #include "api/audio_codecs/audio_encoder.h" |
| #include "api/audio_codecs/audio_encoder_factory.h" |
| #include "api/audio_codecs/audio_encoder_factory_template.h" |
| #include "api/audio_codecs/audio_format.h" |
| #include "api/audio_codecs/opus_audio_decoder_factory.h" |
| #include "api/audio_codecs/opus_audio_encoder_factory.h" |
| #include "api/audio_options.h" |
| #include "api/data_channel_interface.h" |
| #include "api/environment/environment.h" |
| #include "api/media_stream_interface.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/rtc_error.h" |
| #include "api/scoped_refptr.h" |
| #include "media/sctp/sctp_transport_internal.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/physical_socket_server.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| #include "rtc_base/thread.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| #ifdef WEBRTC_ANDROID |
| #include "pc/test/android_test_initializer.h" |
| #endif |
| #include "pc/test/peer_connection_test_wrapper.h" |
| // Notice that mockpeerconnectionobservers.h must be included after the above! |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "test/mock_audio_decoder.h" |
| #include "test/mock_audio_decoder_factory.h" |
| #include "test/mock_audio_encoder_factory.h" |
| |
| using ::testing::_; |
| using ::testing::AtLeast; |
| using ::testing::Invoke; |
| using ::testing::StrictMock; |
| using ::testing::Values; |
| using ::webrtc::DataChannelInterface; |
| using ::webrtc::Environment; |
| using ::webrtc::MediaStreamInterface; |
| using ::webrtc::PeerConnectionInterface; |
| using ::webrtc::SdpSemantics; |
| |
| namespace { |
| |
| const int kMaxWait = 25000; |
| |
| } // namespace |
| |
| class PeerConnectionEndToEndBaseTest : public sigslot::has_slots<>, |
| public ::testing::Test { |
| public: |
| typedef std::vector<rtc::scoped_refptr<DataChannelInterface>> DataChannelList; |
| |
| explicit PeerConnectionEndToEndBaseTest(SdpSemantics sdp_semantics) |
| : network_thread_(std::make_unique<rtc::Thread>(&pss_)), |
| worker_thread_(rtc::Thread::Create()) { |
| RTC_CHECK(network_thread_->Start()); |
| RTC_CHECK(worker_thread_->Start()); |
| caller_ = rtc::make_ref_counted<PeerConnectionTestWrapper>( |
| "caller", &pss_, network_thread_.get(), worker_thread_.get()); |
| callee_ = rtc::make_ref_counted<PeerConnectionTestWrapper>( |
| "callee", &pss_, network_thread_.get(), worker_thread_.get()); |
| webrtc::PeerConnectionInterface::IceServer ice_server; |
| ice_server.uri = "stun:stun.l.google.com:19302"; |
| config_.servers.push_back(ice_server); |
| config_.sdp_semantics = sdp_semantics; |
| |
| #ifdef WEBRTC_ANDROID |
| webrtc::InitializeAndroidObjects(); |
| #endif |
| } |
| |
| void CreatePcs( |
| rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory1, |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory1, |
| rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory2, |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory2) { |
| EXPECT_TRUE(caller_->CreatePc(config_, audio_encoder_factory1, |
| audio_decoder_factory1)); |
| EXPECT_TRUE(callee_->CreatePc(config_, audio_encoder_factory2, |
| audio_decoder_factory2)); |
| PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get()); |
| |
| caller_->SignalOnDataChannel.connect( |
| this, &PeerConnectionEndToEndBaseTest::OnCallerAddedDataChanel); |
| callee_->SignalOnDataChannel.connect( |
| this, &PeerConnectionEndToEndBaseTest::OnCalleeAddedDataChannel); |
| } |
| |
| void CreatePcs( |
| rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) { |
| CreatePcs(audio_encoder_factory, audio_decoder_factory, |
| audio_encoder_factory, audio_decoder_factory); |
| } |
| |
| void GetAndAddUserMedia() { |
| cricket::AudioOptions audio_options; |
| GetAndAddUserMedia(true, audio_options, true); |
| } |
| |
| void GetAndAddUserMedia(bool audio, |
| const cricket::AudioOptions& audio_options, |
| bool video) { |
| caller_->GetAndAddUserMedia(audio, audio_options, video); |
| callee_->GetAndAddUserMedia(audio, audio_options, video); |
| } |
| |
| void Negotiate() { |
| caller_->CreateOffer( |
| webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); |
| } |
| |
| void WaitForCallEstablished() { |
| caller_->WaitForCallEstablished(); |
| callee_->WaitForCallEstablished(); |
| } |
| |
| void WaitForConnection() { |
| caller_->WaitForConnection(); |
| callee_->WaitForConnection(); |
| } |
| |
| void OnCallerAddedDataChanel(DataChannelInterface* dc) { |
| caller_signaled_data_channels_.push_back( |
| rtc::scoped_refptr<DataChannelInterface>(dc)); |
| } |
| |
| void OnCalleeAddedDataChannel(DataChannelInterface* dc) { |
| callee_signaled_data_channels_.push_back( |
| rtc::scoped_refptr<DataChannelInterface>(dc)); |
| } |
| |
| // Tests that `dc1` and `dc2` can send to and receive from each other. |
| void TestDataChannelSendAndReceive(DataChannelInterface* dc1, |
| DataChannelInterface* dc2, |
| size_t size = 6) { |
| std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer( |
| new webrtc::MockDataChannelObserver(dc1)); |
| |
| std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer( |
| new webrtc::MockDataChannelObserver(dc2)); |
| |
| static const std::string kDummyData = |
| "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/"; |
| webrtc::DataBuffer buffer(""); |
| |
| size_t sizeLeft = size; |
| while (sizeLeft > 0) { |
| size_t chunkSize = |
| sizeLeft > kDummyData.length() ? kDummyData.length() : sizeLeft; |
| buffer.data.AppendData(kDummyData.data(), chunkSize); |
| sizeLeft -= chunkSize; |
| } |
| |
| EXPECT_TRUE(dc1->Send(buffer)); |
| EXPECT_EQ_WAIT(buffer.data, |
| rtc::CopyOnWriteBuffer(dc2_observer->last_message()), |
| kMaxWait); |
| |
| EXPECT_TRUE(dc2->Send(buffer)); |
| EXPECT_EQ_WAIT(buffer.data, |
| rtc::CopyOnWriteBuffer(dc1_observer->last_message()), |
| kMaxWait); |
| |
| EXPECT_EQ(1U, dc1_observer->received_message_count()); |
| EXPECT_EQ(size, dc1_observer->last_message().length()); |
| EXPECT_EQ(1U, dc2_observer->received_message_count()); |
| EXPECT_EQ(size, dc2_observer->last_message().length()); |
| } |
| |
| void WaitForDataChannelsToOpen(DataChannelInterface* local_dc, |
| const DataChannelList& remote_dc_list, |
| size_t remote_dc_index) { |
| EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait); |
| |
| ASSERT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait); |
| EXPECT_EQ_WAIT(DataChannelInterface::kOpen, |
| remote_dc_list[remote_dc_index]->state(), kMaxWait); |
| EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id()); |
| } |
| |
| void CloseDataChannels(DataChannelInterface* local_dc, |
| const DataChannelList& remote_dc_list, |
| size_t remote_dc_index) { |
| local_dc->Close(); |
| EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait); |
| EXPECT_EQ_WAIT(DataChannelInterface::kClosed, |
| remote_dc_list[remote_dc_index]->state(), kMaxWait); |
| } |
| |
| protected: |
| rtc::AutoThread main_thread_; |
| rtc::PhysicalSocketServer pss_; |
| std::unique_ptr<rtc::Thread> network_thread_; |
| std::unique_ptr<rtc::Thread> worker_thread_; |
| rtc::scoped_refptr<PeerConnectionTestWrapper> caller_; |
| rtc::scoped_refptr<PeerConnectionTestWrapper> callee_; |
| DataChannelList caller_signaled_data_channels_; |
| DataChannelList callee_signaled_data_channels_; |
| webrtc::PeerConnectionInterface::RTCConfiguration config_; |
| }; |
| |
| class PeerConnectionEndToEndTest |
| : public PeerConnectionEndToEndBaseTest, |
| public ::testing::WithParamInterface<SdpSemantics> { |
| protected: |
| PeerConnectionEndToEndTest() : PeerConnectionEndToEndBaseTest(GetParam()) {} |
| }; |
| |
| namespace { |
| |
| std::unique_ptr<webrtc::AudioDecoder> CreateForwardingMockDecoder( |
| std::unique_ptr<webrtc::AudioDecoder> real_decoder) { |
| class ForwardingMockDecoder : public StrictMock<webrtc::MockAudioDecoder> { |
| public: |
| explicit ForwardingMockDecoder(std::unique_ptr<AudioDecoder> decoder) |
| : decoder_(std::move(decoder)) {} |
| |
| private: |
| std::unique_ptr<AudioDecoder> decoder_; |
| }; |
| |
| const auto dec = real_decoder.get(); // For lambda capturing. |
| auto mock_decoder = |
| std::make_unique<ForwardingMockDecoder>(std::move(real_decoder)); |
| EXPECT_CALL(*mock_decoder, Channels()) |
| .Times(AtLeast(1)) |
| .WillRepeatedly(Invoke([dec] { return dec->Channels(); })); |
| EXPECT_CALL(*mock_decoder, DecodeInternal(_, _, _, _, _)) |
| .Times(AtLeast(1)) |
| .WillRepeatedly( |
| Invoke([dec](const uint8_t* encoded, size_t encoded_len, |
| int sample_rate_hz, int16_t* decoded, |
| webrtc::AudioDecoder::SpeechType* speech_type) { |
| return dec->Decode(encoded, encoded_len, sample_rate_hz, |
| std::numeric_limits<size_t>::max(), decoded, |
| speech_type); |
| })); |
| EXPECT_CALL(*mock_decoder, Die()); |
| EXPECT_CALL(*mock_decoder, HasDecodePlc()).WillRepeatedly(Invoke([dec] { |
| return dec->HasDecodePlc(); |
| })); |
| EXPECT_CALL(*mock_decoder, PacketDuration(_, _)) |
| .Times(AtLeast(1)) |
| .WillRepeatedly(Invoke([dec](const uint8_t* encoded, size_t encoded_len) { |
| return dec->PacketDuration(encoded, encoded_len); |
| })); |
| EXPECT_CALL(*mock_decoder, SampleRateHz()) |
| .Times(AtLeast(1)) |
| .WillRepeatedly(Invoke([dec] { return dec->SampleRateHz(); })); |
| |
| return std::move(mock_decoder); |
| } |
| |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> |
| CreateForwardingMockDecoderFactory( |
| webrtc::AudioDecoderFactory* real_decoder_factory) { |
| rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory = |
| rtc::make_ref_counted<StrictMock<webrtc::MockAudioDecoderFactory>>(); |
| EXPECT_CALL(*mock_decoder_factory, GetSupportedDecoders()) |
| .Times(AtLeast(1)) |
| .WillRepeatedly(Invoke([real_decoder_factory] { |
| return real_decoder_factory->GetSupportedDecoders(); |
| })); |
| EXPECT_CALL(*mock_decoder_factory, IsSupportedDecoder(_)) |
| .Times(AtLeast(1)) |
| .WillRepeatedly( |
| Invoke([real_decoder_factory](const webrtc::SdpAudioFormat& format) { |
| return real_decoder_factory->IsSupportedDecoder(format); |
| })); |
| EXPECT_CALL(*mock_decoder_factory, Create) |
| .Times(AtLeast(2)) |
| .WillRepeatedly( |
| [real_decoder_factory]( |
| const webrtc::Environment& env, |
| const webrtc::SdpAudioFormat& format, |
| std::optional<webrtc::AudioCodecPairId> codec_pair_id) { |
| auto real_decoder = |
| real_decoder_factory->Create(env, format, codec_pair_id); |
| return real_decoder |
| ? CreateForwardingMockDecoder(std::move(real_decoder)) |
| : nullptr; |
| }); |
| return mock_decoder_factory; |
| } |
| |
| struct AudioEncoderUnicornSparklesRainbow { |
| using Config = webrtc::AudioEncoderL16::Config; |
| static std::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) { |
| if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) { |
| const webrtc::CodecParameterMap expected_params = {{"num_horns", "1"}}; |
| EXPECT_EQ(expected_params, format.parameters); |
| format.parameters.clear(); |
| format.name = "L16"; |
| return webrtc::AudioEncoderL16::SdpToConfig(format); |
| } else { |
| return std::nullopt; |
| } |
| } |
| static void AppendSupportedEncoders( |
| std::vector<webrtc::AudioCodecSpec>* specs) { |
| std::vector<webrtc::AudioCodecSpec> new_specs; |
| webrtc::AudioEncoderL16::AppendSupportedEncoders(&new_specs); |
| for (auto& spec : new_specs) { |
| spec.format.name = "UnicornSparklesRainbow"; |
| EXPECT_TRUE(spec.format.parameters.empty()); |
| spec.format.parameters.emplace("num_horns", "1"); |
| specs->push_back(spec); |
| } |
| } |
| static webrtc::AudioCodecInfo QueryAudioEncoder(const Config& config) { |
| return webrtc::AudioEncoderL16::QueryAudioEncoder(config); |
| } |
| static std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder( |
| const Config& config, |
| int payload_type, |
| std::optional<webrtc::AudioCodecPairId> codec_pair_id = std::nullopt) { |
| return webrtc::AudioEncoderL16::MakeAudioEncoder(config, payload_type, |
| codec_pair_id); |
| } |
| }; |
| |
| struct AudioDecoderUnicornSparklesRainbow { |
| using Config = webrtc::AudioDecoderL16::Config; |
| static std::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) { |
| if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) { |
| const webrtc::CodecParameterMap expected_params = {{"num_horns", "1"}}; |
| EXPECT_EQ(expected_params, format.parameters); |
| format.parameters.clear(); |
| format.name = "L16"; |
| return webrtc::AudioDecoderL16::SdpToConfig(format); |
| } else { |
| return std::nullopt; |
| } |
| } |
| static void AppendSupportedDecoders( |
| std::vector<webrtc::AudioCodecSpec>* specs) { |
| std::vector<webrtc::AudioCodecSpec> new_specs; |
| webrtc::AudioDecoderL16::AppendSupportedDecoders(&new_specs); |
| for (auto& spec : new_specs) { |
| spec.format.name = "UnicornSparklesRainbow"; |
| EXPECT_TRUE(spec.format.parameters.empty()); |
| spec.format.parameters.emplace("num_horns", "1"); |
| specs->push_back(spec); |
| } |
| } |
| static std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder( |
| const Config& config, |
| std::optional<webrtc::AudioCodecPairId> codec_pair_id = std::nullopt) { |
| return webrtc::AudioDecoderL16::MakeAudioDecoder(config, codec_pair_id); |
| } |
| }; |
| |
| } // namespace |
| |
| TEST_P(PeerConnectionEndToEndTest, Call) { |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory = |
| webrtc::CreateOpusAudioDecoderFactory(); |
| CreatePcs(webrtc::CreateOpusAudioEncoderFactory(), |
| CreateForwardingMockDecoderFactory(real_decoder_factory.get())); |
| GetAndAddUserMedia(); |
| Negotiate(); |
| WaitForCallEstablished(); |
| } |
| |
| #if defined(IS_FUCHSIA) |
| TEST_P(PeerConnectionEndToEndTest, CallWithSdesKeyNegotiation) { |
| config_.enable_dtls_srtp = false; |
| CreatePcs(webrtc::CreateOpusAudioEncoderFactory(), |
| webrtc::CreateOpusAudioDecoderFactory()); |
| GetAndAddUserMedia(); |
| Negotiate(); |
| WaitForCallEstablished(); |
| } |
| #endif |
| |
| TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) { |
| class IdLoggingAudioEncoderFactory : public webrtc::AudioEncoderFactory { |
| public: |
| IdLoggingAudioEncoderFactory( |
| rtc::scoped_refptr<AudioEncoderFactory> real_factory, |
| std::vector<webrtc::AudioCodecPairId>* const codec_ids) |
| : fact_(real_factory), codec_ids_(codec_ids) {} |
| std::vector<webrtc::AudioCodecSpec> GetSupportedEncoders() override { |
| return fact_->GetSupportedEncoders(); |
| } |
| std::optional<webrtc::AudioCodecInfo> QueryAudioEncoder( |
| const webrtc::SdpAudioFormat& format) override { |
| return fact_->QueryAudioEncoder(format); |
| } |
| std::unique_ptr<webrtc::AudioEncoder> Create( |
| const Environment& env, |
| const webrtc::SdpAudioFormat& format, |
| Options options) override { |
| EXPECT_TRUE(options.codec_pair_id.has_value()); |
| codec_ids_->push_back(*options.codec_pair_id); |
| return fact_->Create(env, format, options); |
| } |
| |
| private: |
| const rtc::scoped_refptr<webrtc::AudioEncoderFactory> fact_; |
| std::vector<webrtc::AudioCodecPairId>* const codec_ids_; |
| }; |
| |
| class IdLoggingAudioDecoderFactory : public webrtc::AudioDecoderFactory { |
| public: |
| IdLoggingAudioDecoderFactory( |
| rtc::scoped_refptr<AudioDecoderFactory> real_factory, |
| std::vector<webrtc::AudioCodecPairId>* const codec_ids) |
| : fact_(real_factory), codec_ids_(codec_ids) {} |
| std::vector<webrtc::AudioCodecSpec> GetSupportedDecoders() override { |
| return fact_->GetSupportedDecoders(); |
| } |
| bool IsSupportedDecoder(const webrtc::SdpAudioFormat& format) override { |
| return fact_->IsSupportedDecoder(format); |
| } |
| std::unique_ptr<webrtc::AudioDecoder> Create( |
| const Environment& env, |
| const webrtc::SdpAudioFormat& format, |
| std::optional<webrtc::AudioCodecPairId> codec_pair_id) override { |
| EXPECT_TRUE(codec_pair_id.has_value()); |
| codec_ids_->push_back(*codec_pair_id); |
| return fact_->Create(env, format, codec_pair_id); |
| } |
| |
| private: |
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory> fact_; |
| std::vector<webrtc::AudioCodecPairId>* const codec_ids_; |
| }; |
| |
| std::vector<webrtc::AudioCodecPairId> encoder_id1, encoder_id2, decoder_id1, |
| decoder_id2; |
| CreatePcs(rtc::make_ref_counted<IdLoggingAudioEncoderFactory>( |
| webrtc::CreateAudioEncoderFactory< |
| AudioEncoderUnicornSparklesRainbow>(), |
| &encoder_id1), |
| rtc::make_ref_counted<IdLoggingAudioDecoderFactory>( |
| webrtc::CreateAudioDecoderFactory< |
| AudioDecoderUnicornSparklesRainbow>(), |
| &decoder_id1), |
| rtc::make_ref_counted<IdLoggingAudioEncoderFactory>( |
| webrtc::CreateAudioEncoderFactory< |
| AudioEncoderUnicornSparklesRainbow>(), |
| &encoder_id2), |
| rtc::make_ref_counted<IdLoggingAudioDecoderFactory>( |
| webrtc::CreateAudioDecoderFactory< |
| AudioDecoderUnicornSparklesRainbow>(), |
| &decoder_id2)); |
| GetAndAddUserMedia(); |
| Negotiate(); |
| WaitForCallEstablished(); |
| |
| // Each codec factory has been used to create one codec. The first pair got |
| // the same ID because they were passed to the same PeerConnectionFactory, |
| // and the second pair got the same ID---but these two IDs are not equal, |
| // because each PeerConnectionFactory has its own ID. |
| EXPECT_EQ(1U, encoder_id1.size()); |
| EXPECT_EQ(1U, encoder_id2.size()); |
| EXPECT_EQ(encoder_id1, decoder_id1); |
| EXPECT_EQ(encoder_id2, decoder_id2); |
| EXPECT_NE(encoder_id1, encoder_id2); |
| } |
| |
| #ifdef WEBRTC_HAVE_SCTP |
| // Verifies that a DataChannel created before the negotiation can transition to |
| // "OPEN" and transfer data. |
| TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { |
| CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(), |
| webrtc::MockAudioDecoderFactory::CreateEmptyFactory()); |
| |
| webrtc::DataChannelInit init; |
| rtc::scoped_refptr<DataChannelInterface> caller_dc( |
| caller_->CreateDataChannel("data", init)); |
| rtc::scoped_refptr<DataChannelInterface> callee_dc( |
| callee_->CreateDataChannel("data", init)); |
| |
| Negotiate(); |
| WaitForConnection(); |
| |
| WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0); |
| WaitForDataChannelsToOpen(callee_dc.get(), caller_signaled_data_channels_, 0); |
| |
| TestDataChannelSendAndReceive(caller_dc.get(), |
| callee_signaled_data_channels_[0].get()); |
| TestDataChannelSendAndReceive(callee_dc.get(), |
| caller_signaled_data_channels_[0].get()); |
| |
| CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 0); |
| CloseDataChannels(callee_dc.get(), caller_signaled_data_channels_, 0); |
| } |
| |
| // Verifies that a DataChannel created after the negotiation can transition to |
| // "OPEN" and transfer data. |
| TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) { |
| CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(), |
| webrtc::MockAudioDecoderFactory::CreateEmptyFactory()); |
| |
| webrtc::DataChannelInit init; |
| |
| // This DataChannel is for creating the data content in the negotiation. |
| rtc::scoped_refptr<DataChannelInterface> dummy( |
| caller_->CreateDataChannel("data", init)); |
| Negotiate(); |
| WaitForConnection(); |
| |
| // Wait for the data channel created pre-negotiation to be opened. |
| WaitForDataChannelsToOpen(dummy.get(), callee_signaled_data_channels_, 0); |
| |
| // Create new DataChannels after the negotiation and verify their states. |
| rtc::scoped_refptr<DataChannelInterface> caller_dc( |
| caller_->CreateDataChannel("hello", init)); |
| rtc::scoped_refptr<DataChannelInterface> callee_dc( |
| callee_->CreateDataChannel("hello", init)); |
| |
| WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1); |
| WaitForDataChannelsToOpen(callee_dc.get(), caller_signaled_data_channels_, 0); |
| |
| TestDataChannelSendAndReceive(caller_dc.get(), |
| callee_signaled_data_channels_[1].get()); |
| TestDataChannelSendAndReceive(callee_dc.get(), |
| caller_signaled_data_channels_[0].get()); |
| |
| CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1); |
| CloseDataChannels(callee_dc.get(), caller_signaled_data_channels_, 0); |
| } |
| |
| // Verifies that a DataChannel created can transfer large messages. |
| TEST_P(PeerConnectionEndToEndTest, CreateDataChannelLargeTransfer) { |
| CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(), |
| webrtc::MockAudioDecoderFactory::CreateEmptyFactory()); |
| |
| webrtc::DataChannelInit init; |
| |
| // This DataChannel is for creating the data content in the negotiation. |
| rtc::scoped_refptr<DataChannelInterface> dummy( |
| caller_->CreateDataChannel("data", init)); |
| Negotiate(); |
| WaitForConnection(); |
| |
| // Wait for the data channel created pre-negotiation to be opened. |
| WaitForDataChannelsToOpen(dummy.get(), callee_signaled_data_channels_, 0); |
| |
| // Create new DataChannels after the negotiation and verify their states. |
| rtc::scoped_refptr<DataChannelInterface> caller_dc( |
| caller_->CreateDataChannel("hello", init)); |
| rtc::scoped_refptr<DataChannelInterface> callee_dc( |
| callee_->CreateDataChannel("hello", init)); |
| |
| WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1); |
| WaitForDataChannelsToOpen(callee_dc.get(), caller_signaled_data_channels_, 0); |
| |
| TestDataChannelSendAndReceive( |
| caller_dc.get(), callee_signaled_data_channels_[1].get(), 256 * 1024); |
| TestDataChannelSendAndReceive( |
| callee_dc.get(), caller_signaled_data_channels_[0].get(), 256 * 1024); |
| |
| CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1); |
| CloseDataChannels(callee_dc.get(), caller_signaled_data_channels_, 0); |
| } |
| |
| // Verifies that DataChannel IDs are even/odd based on the DTLS roles. |
| TEST_P(PeerConnectionEndToEndTest, DataChannelIdAssignment) { |
| CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(), |
| webrtc::MockAudioDecoderFactory::CreateEmptyFactory()); |
| |
| webrtc::DataChannelInit init; |
| rtc::scoped_refptr<DataChannelInterface> caller_dc_1( |
| caller_->CreateDataChannel("data", init)); |
| rtc::scoped_refptr<DataChannelInterface> callee_dc_1( |
| callee_->CreateDataChannel("data", init)); |
| |
| Negotiate(); |
| WaitForConnection(); |
| |
| EXPECT_EQ(1, caller_dc_1->id() % 2); |
| EXPECT_EQ(0, callee_dc_1->id() % 2); |
| |
| rtc::scoped_refptr<DataChannelInterface> caller_dc_2( |
| caller_->CreateDataChannel("data", init)); |
| rtc::scoped_refptr<DataChannelInterface> callee_dc_2( |
| callee_->CreateDataChannel("data", init)); |
| |
| EXPECT_EQ(1, caller_dc_2->id() % 2); |
| EXPECT_EQ(0, callee_dc_2->id() % 2); |
| } |
| |
| // Verifies that the message is received by the right remote DataChannel when |
| // there are multiple DataChannels. |
| TEST_P(PeerConnectionEndToEndTest, |
| MessageTransferBetweenTwoPairsOfDataChannels) { |
| CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(), |
| webrtc::MockAudioDecoderFactory::CreateEmptyFactory()); |
| |
| webrtc::DataChannelInit init; |
| |
| rtc::scoped_refptr<DataChannelInterface> caller_dc_1( |
| caller_->CreateDataChannel("data", init)); |
| rtc::scoped_refptr<DataChannelInterface> caller_dc_2( |
| caller_->CreateDataChannel("data", init)); |
| |
| Negotiate(); |
| WaitForConnection(); |
| WaitForDataChannelsToOpen(caller_dc_1.get(), callee_signaled_data_channels_, |
| 0); |
| WaitForDataChannelsToOpen(caller_dc_2.get(), callee_signaled_data_channels_, |
| 1); |
| |
| std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer( |
| new webrtc::MockDataChannelObserver( |
| callee_signaled_data_channels_[0].get())); |
| |
| std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer( |
| new webrtc::MockDataChannelObserver( |
| callee_signaled_data_channels_[1].get())); |
| |
| const std::string message_1 = "hello 1"; |
| const std::string message_2 = "hello 2"; |
| |
| caller_dc_1->Send(webrtc::DataBuffer(message_1)); |
| EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait); |
| |
| caller_dc_2->Send(webrtc::DataBuffer(message_2)); |
| EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait); |
| |
| EXPECT_EQ(1U, dc_1_observer->received_message_count()); |
| EXPECT_EQ(1U, dc_2_observer->received_message_count()); |
| } |
| |
| // Verifies that a DataChannel added from an OPEN message functions after |
| // a channel has been previously closed (webrtc issue 3778). |
| // This previously failed because the new channel re-used the ID of the closed |
| // channel, and the closed channel was incorrectly still assigned to the ID. |
| TEST_P(PeerConnectionEndToEndTest, |
| DataChannelFromOpenWorksAfterPreviousChannelClosed) { |
| CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(), |
| webrtc::MockAudioDecoderFactory::CreateEmptyFactory()); |
| |
| webrtc::DataChannelInit init; |
| rtc::scoped_refptr<DataChannelInterface> caller_dc( |
| caller_->CreateDataChannel("data", init)); |
| |
| Negotiate(); |
| WaitForConnection(); |
| |
| WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0); |
| int first_channel_id = caller_dc->id(); |
| // Wait for the local side to say it's closed, but not the remote side. |
| // Previously, the channel on which Close is called reported being closed |
| // prematurely, and this caused issues; see bugs.webrtc.org/4453. |
| caller_dc->Close(); |
| EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait); |
| |
| // Create a new channel and ensure it works after closing the previous one. |
| caller_dc = caller_->CreateDataChannel("data2", init); |
| WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1); |
| // Since the second channel was created after the first finished closing, it |
| // should be able to re-use the first one's ID. |
| EXPECT_EQ(first_channel_id, caller_dc->id()); |
| TestDataChannelSendAndReceive(caller_dc.get(), |
| callee_signaled_data_channels_[1].get()); |
| |
| CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1); |
| } |
| |
| // This tests that if a data channel is closed remotely while not referenced |
| // by the application (meaning only the PeerConnection contributes to its |
| // reference count), no memory access violation will occur. |
| // See: https://code.google.com/p/chromium/issues/detail?id=565048 |
| TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) { |
| CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(), |
| webrtc::MockAudioDecoderFactory::CreateEmptyFactory()); |
| |
| webrtc::DataChannelInit init; |
| rtc::scoped_refptr<DataChannelInterface> caller_dc( |
| caller_->CreateDataChannel("data", init)); |
| |
| Negotiate(); |
| WaitForConnection(); |
| |
| WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0); |
| // This removes the reference to the remote data channel that we hold. |
| callee_signaled_data_channels_.clear(); |
| caller_dc->Close(); |
| EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait); |
| |
| // Wait for a bit longer so the remote data channel will receive the |
| // close message and be destroyed. |
| rtc::Thread::Current()->ProcessMessages(100); |
| } |
| |
| // Test behavior of creating too many datachannels. |
| TEST_P(PeerConnectionEndToEndTest, TooManyDataChannelsOpenedBeforeConnecting) { |
| CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(), |
| webrtc::MockAudioDecoderFactory::CreateEmptyFactory()); |
| |
| webrtc::DataChannelInit init; |
| std::vector<rtc::scoped_refptr<DataChannelInterface>> channels; |
| for (int i = 0; i <= cricket::kMaxSctpStreams / 2; i++) { |
| rtc::scoped_refptr<DataChannelInterface> caller_dc( |
| caller_->CreateDataChannel("data", init)); |
| channels.push_back(std::move(caller_dc)); |
| } |
| Negotiate(); |
| WaitForConnection(); |
| EXPECT_EQ_WAIT(callee_signaled_data_channels_.size(), |
| static_cast<size_t>(cricket::kMaxSctpStreams / 2), kMaxWait); |
| EXPECT_EQ(DataChannelInterface::kOpen, |
| channels[(cricket::kMaxSctpStreams / 2) - 1]->state()); |
| EXPECT_EQ(DataChannelInterface::kClosed, |
| channels[cricket::kMaxSctpStreams / 2]->state()); |
| } |
| |
| #endif // WEBRTC_HAVE_SCTP |
| |
| TEST_P(PeerConnectionEndToEndTest, CanRestartIce) { |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory = |
| webrtc::CreateOpusAudioDecoderFactory(); |
| CreatePcs(webrtc::CreateOpusAudioEncoderFactory(), |
| CreateForwardingMockDecoderFactory(real_decoder_factory.get())); |
| GetAndAddUserMedia(); |
| Negotiate(); |
| WaitForCallEstablished(); |
| // Cause ICE restart to be requested. |
| auto config = caller_->pc()->GetConfiguration(); |
| ASSERT_NE(PeerConnectionInterface::kRelay, config.type); |
| config.type = PeerConnectionInterface::kRelay; |
| ASSERT_TRUE(caller_->pc()->SetConfiguration(config).ok()); |
| // When solving https://crbug.com/webrtc/10504, all we need to check |
| // is that we do not crash. We should also be testing that restart happens. |
| } |
| |
| INSTANTIATE_TEST_SUITE_P(PeerConnectionEndToEndTest, |
| PeerConnectionEndToEndTest, |
| Values(SdpSemantics::kPlanB_DEPRECATED, |
| SdpSemantics::kUnifiedPlan)); |
